Check that supervisor cannot connect to recording server if right is not allowed

Warning: This test is supposed to be not required as we have the daily install automated test

Check that the XiVO Edge FQDN field in webi is emptied when upgrading from a version under 2021.07.00

This tests check if the user list API works as intended

The aim of this test is to prevent import of users with same usernames.

• duplicateusername.csv
• duplicateusername.csv

The aim of this test is to prevent import of users if the CSV file has unknown or mistyped header column names.

• wrong_header.csv
• wrong_header.csv

The purpose of this test is to verify that users without cti profile can be updated with cti username and password in the csv

CSV import/update continues in the background after the timeout error

The goal is to check that user receives a warning when the password does not comply with policy check

• import_csv_password : import_password.csv
• import_password.csv

This test checks that the password policy notice displays correctly

Objectif: Il doit être possible de changer le device d'un utilisateur en une seule opération dans la webi.

Objectif: Il doit être possible de déassocié un utilisateur d'un poste SCCP dans la webi

Purpose: Caller id and connected line id are correct when SCCP phones are used to call SCCP and/or SIP

Lorsqu'un fichier audio est joué pas asterisk, il doit être dans la même langue que celle choisie pour l'utilisateur associé, ou dans la langue des Paramètres Généraux par défaut.

Un téléphone SCCP peut faire des transferts directs et indirects

On test la musique d'attente

S'assurer que le early-rtp fonctionne avec un poste SCCP

Un appel vers nous est ensuite raccroché sans qu'on l'ait pris, tout ca pendant qu'on était en ligne.

If the SCCP direct media is active and the SIP direct media is not active, both sides of the communication must hear sound.

Si le direct media est actif sur SCCP et que j'effectue un appel A appel B renvoyé vers C direct media devrait être utilisé

Si le direct media SCCP est actif et qu'un agent utilise un téléphone SCCP, les clients qui appellent peuvent entendre l'agent et vice-versa

Si le direct media est actif sur SCCP et que je mets en pause et reprend un appel, alors le direct media devrait encore être utilisé.

Si la personne transféré raccroche pendant que le transfert est en cours, le "transféreur" et l'appelé sont toujours en communication.

Objectif: Il doit être possible de synchroniser des postes SCCP depuis l'interface web.

Un utilisateur avec un device SCCP peut faire du group pickup

Recording saves call records and Configuration manages users that can access it

Agen can login, change styate, see available agents on its queue

Checks the Kibana panels

Checks SpagoBI and statistics using sample reports

Validate that the call history is generated and available through xuc interfaces

Verify that desktop assistant is available to be downbloaded from web interface and at least runs on Windows

Verify that desktop assistant MSI installer is build and works

Test an outgoing WebRTC call using the latest Stable and Beta release of Chrome under Linux.

Please, write down in the comment box the exact versions of Chrome tested.

Test an outgoing WebRTC call using the latest Stable and Beta release of Chrome under Windows.

Please, write down in the comment box the exact versions of Chrome tested.

Test an outgoing WebRTC call using the latest Beta release of Chrome on the oldest currently supported LTS (at the time of writing -2020/12- this is Aldebaran).

Please, write down in the comment box the exact versions of Chrome tested.

The goal is to check that Chrome didn't deprecate any WebRTC and/or SSL/TLS that won't be compatible with an older version/configuration of asterisk (the goal is to not reproduce the #3676 bug)

For Aldebaran please follow procedure in ticket https://projects.xivo.solutions/issues/3676#Fix

Webrtc user can hangup a second call from the modal or from the call line

Organizers can invite a user to the conference

participants in a conference can start a second call by typing a phone number in the search bar 

Organizers can invite to conference by typing a phone number in the search bar

Tests the search bar display and actions : call, invite to conference and search

This tests aims to display A flashtext received to a connected User in UC assistant

This test checks if you can disable chat functionality using custom.env

This test aims to check if :

- A toast message appears when a user cannot receive a flashtext

- The css for the long usernames is ok 

Be able to retrieve chat history between U1 and U2

Be able to send a chat message while other party is offline

Verify that user receives unread notification after log in

To verify if a user has unread messages, opening the chat will mark them as read

Verify that chat message can be sent to every xivo user.

Verify that a chat message can be send to a user who was never logged in

Verify that users can continue send and receive messages in case Asterisk/cti link crashed and is restored

Test Single sign-on feature with OIDC Server

On incomming call ringing, Display caller name in modal, notification. After acceptimng call display it in call list.

Test that notification and a popup are displayed when a long call is put On Hold

When a yealink user receives a second call, he is not able to answer from the uc assistant, but he can hangup the call. 

Email can be sent without pre configured template

The purpose of this test is to verify you can set a template for email sending, corretly fill variable in the template and set the mailTo link accordingly.

Search with less than two characters is prevented and triggers a tooltip 

This test verifies that when a user leaves the UC/Desktop assistant, even by not using the logout button, if that user has a call dialing/ringing or ongoing, this call will correctly be shut down without waiting for RTP timeout.

It is related to #3878

phone users other than yealink can answer and hangup second calls 

User should see a letter icon when receiving a message

During conversation the letter is not displayed if the message is for this conversation

Manager can login, see queues, states of agents, calls and other activity at call center.

check that the external user phone number is well displayed in ccagent

Test that notification and a popup are displayed when a long call is put On Hold

Check that the call line is displayed correclty and the volume meter works for webrtc

Test that only agent with correct line setup can connect.

Test conference from XiVO UC assistant

As a webrtc user, while I'm in a call, I want to be able to ignore an incoming call.

Tests Recording API to listen recorded calls 

Check if the electron tray icon gets updated correctly on missed calls and on new flashtext

Test the global installation (and not userland specific) for electron application on Windows

This test is to check if we can correctly disable the download buttons that are on the login page

This test aims to verify that a webrtc user on mds1 can call and receive internal calls from a phone user on MDS0.  

Calls between webrtc users on different MDS

Calls between external user and webrtc on MDS 

I can call a Group/Queue/Meetme configured on XiVO Main

Test that dialplans, sounds and music on hold is synchronized between all XDS servers

• mohpiano.doc
• mohpiano.doc

The purpose of this test is to check you can edit your stun/turn server configuration and propagate it to the asterisk

The purpose of this test is to check you can edit your stun/turn server configuration and propagate it to the web applications.

The purpose of this test is to verify the transport type for webrtc peers is

transport = ws without stun server defined

transport = udp with stun server defined